EP1762120B1 - Audio conferencing device, audio conferencing system and method for carrying out an audio conference - Google Patents

Abstract

A voice conference apparatus consists of at least one base conference block having three or more signal ports, each including an input for a digitalized voice signal and an output for a digitalized voice signal, wherein a base conference block is configured such that, at a signal port, optionally a terminal or another base conference block may be coupled up using a coupler for the realization of a voice conference apparatus extended with respect to the number of participants or in terms of space, and wherein a base conference block is configured to enable a voice conference among the terminals or base conference blocks coupled thereto, and a remote coupler configured to couple a base conference block with at least one other spatially remote base conference block in order to enable a spatially extended voice conference. A voice conference system may be composed of at least two spatially separate voice conference apparatuses and is structured in a decentralized and modular way and is flexibly extensible.

Description

The present invention generally relates to a voice conference apparatus and method for implementing a voice conference, more particularly to a distributed digital wireless voice conference system.

Radio-based voice conference systems are used wherever communication within a group of people is not possible directly due to distance, high noise levels or other reasons. Several people should be able to talk to each other over wirelessly connected headsets, whereby everyone can hear all other participants at any time. That is, each subscriber has a duplex voice connection.

In a modern voice conference system, a quasi-delay-free and fully duplex-capable voice connection should be established between a large number of listeners. It is important for economic reasons to keep the installation costs as low as possible. Furthermore, it is very important that the system is modular and flexible to a system of virtually any size can be extended. In addition, existing resources should be able to continue to be used. On the customer side, this particularly affects already existing cabling as well as end devices. On the part of the manufacturer already existing circuit concepts will be used. The circuit technology used should be easy to implement at low cost. Excellent voice quality is also expected today from a modern voice conference system. In particular, interference signals coupled in on the transmission link and in a conference node must be very small. After all For wireless systems it is necessary to use the available radio resources as effectively as possible. The sum of these requirements necessitates the further development of the existing concepts for voice conference systems.

In the prior art, voice conference systems with analog-coupled base stations or wireless DECT private branch exchanges are commonly used today. In conference systems with analogically coupled base stations, the handsets connected to a base station are in conference. Several base stations are coupled in the same way. The circuitry needed for conference building includes analog circuitry, which should prevent the coupling of interference in the analog signals used with great effort. This requires elaborate filters and is not ideally possible. With larger numbers of participants, the noise would add up to the uselessness of the system. Conference systems with analogically coupled base stations thus do not meet the high demands placed on modern voice conference systems.

The installation effort is very high due to the need to use well shielded cables. Even with a small number of participants, the voice quality is not satisfactory and interference can be easily coupled. Similarly, the flexible extension to a large number of participants due to the problem of interference is not easily possible. Finally, the use of analog circuit technology today brings significantly higher costs compared to digital circuits.

In addition to the above-described conference systems with analog-coupled base stations today wireless DECT private branch exchanges with digital voice transmission are common. The conference call takes place here in a central unit. This also brings with it some disadvantages. Especially The use of a central processing unit that can sustain a large number of radio links simultaneously is very costly. Modular expansion of the system is difficult. Likewise, the use of a central unit entails a spatial restriction of the conference system. Finally, the internal structure of a central unit differs greatly from that of a base station of a normal, ie non-conferenceable, cordless telephone system. This leads to high costs.

The document DE 10113088 A1 discloses a conference room sound system having a base conference block having a plurality of signal ports each having an input and an output for a speech signal.

The realization of a conference circuit within a central unit is conventionally carried out by using a signal processor which is designed to carry out the calculations necessary for the realization of a speech conference block. Such a signal processor is for example the type SC14428 from National Semiconductor. For further details on the capabilities of the signal processor, please refer to the corresponding documentation.

Based on this prior art, the present invention has for its object to provide a modular digital voice conference apparatus, a voice conference system and a method for operating the same.

This object is achieved by a voice conference apparatus according to claim 1, a voice conference system according to claim 13 and by a method according to claim 29.

The present invention provides a voice conference apparatus having at least one base conference block having three or more signal ports, each having an input for a digitized voice signal and an output for a digitized voice signal, wherein a base conference block is adapted to be connected to a signal port with the aid of a coupling device optionally a terminal or another base conference block to realize a respect to the number of participants or a spatially enhanced voice conference apparatus, and wherein a base conference block is arranged to allow a voice conference between the terminals or base conference blocks coupled thereto and a remote unit adapted to receive a base conference block with at least one further remote Pair the basic conference block to allow a room-to-office voice conference.

In addition, the present invention provides a voice conference system and method for operating a voice conference system.

The present invention is based on the recognition that in a fully digital realization of the speech signal processing and transmission ensures a high voice quality, effectively suppressing interference and system costs can be reduced by using digital circuitry. Furthermore, the present invention is based on the recognition that by using flexibly connectable basic conference blocks with little effort, a modular, virtually arbitrarily expandable voice conference system can be constructed. A decentralized formation of sub-conferences, which is made possible by the fact that spatially separated basic conference blocks can be coupled, makes a complex central unit superfluous. Individual nodes of the conference system differ only in the software, but not in the hardware.

The transmission of the digital voice signals between multiple basic conference blocks, which is enabled by the use of a remote switch allows a spatially enhanced voice conference with high voice quality, wherein a single base conference block calculates a local sub-conference, which leads to a distribution of computational effort and the reduced amount of data compared to conventional centralized conference systems.

Thus, the present voice conference system deviates from the systems known from the prior art in particular in that voice transmission and voice processing are entirely digital, without the need for a central station defined by hardware features. Rather, much of the signal processing is decentralized. Such a voice conference device or a voice conference system brings a number of advantages with it. In particular, voice quality is high due to digital processing and transmission, and susceptibility to interference is low. The distributed implementation reduces the hardware and installation costs, since the wiring significantly simplifies compared to conventional systems. A flexible extension is ensured by the modularity. A node in such a distributed voice conference system differs little from the hardware in a normal telephone system, except for a spatial extension enabling remote coupling device, so that existing hardware can be used with only minor modifications. It is possible to assemble a voice conference system from a plurality of hardware essentially identical spatially distributed units.

In one embodiment of the present speech conferencing apparatus, a base conference block is designed so that at each output of a signal gate a digitally formed sum signal of the input signals of the respective other signal gates is applied in digital form. This has the advantage that a very simple summation is possible by the digital processing. Furthermore, a signal reflection, ie the output of the read from the input of a signal signal can be suppressed signal to the output of the same signal port. Finally, such a design of the basic conference block allows any expansion of the system, wherein only digital signals must be transmitted on all links.

In a further preferred embodiment, one or more of the terminals are coupled via a wireless connection to a base conference block, which is made possible by a suitable design of the remote coupling device. In particular, the wireless connection to one or more terminals according to the DECT standard or a standard based thereon can be done. The wireless connection has the advantage that the conference participants can move freely in the room. The transmission of voice signals according to a digital standard in this case allows a high voice quality with good suppression of interference and good use of the available frequency resources. The conversion of the audio signals into digital signals and vice versa takes place directly in the terminal. If, in addition, the wireless connection is made in accordance with the DECT standard or a standard based thereon, existing or cost-effective standard terminals and base stations can be used.

In another embodiment, a listening and speaking station is coupled via a device for converting analog signals into digital signals and a device for converting digital signals into analog signals to a base conference block. This allows the operation of a listening and speaking station, which is located in the same location as the basic conference block. Thus, with low cost, another stationary user can be included in the speech conference.

In another embodiment, one or more further base conference blocks are coupled via a wireless connection to a first base conference block. This has the advantage that the voice conference device can be extended without any installation effort. Such a solution also offers the advantage that the available frequency resources can be used effectively. The transmission power of the handsets need only be large enough to reach the nearest base conference block, while the optionally further distant base conference blocks communicate with one another over a radio link with greater transmission power.

In another preferred embodiment, one or more further basic conference blocks are coupled via a wired bus to a first base conference block. Again, the remote coupling task is to enable and manage the connection. In particular, it is advantageous if the wired bus is designed for the transmission of multiple voice channels in time division multiplex. Such an arrangement has the advantage that a voice conference device can be arbitrarily expanded spatially, without occupying frequency resources. With the help of a wired bus almost any distance can be bridged. In particular, the transition to long-distance telephone networks is possible. The transmission of multiple voice channels in time division multiplex on a bus makes it possible in an advantageous manner to produce a plurality of connections with low installation costs. The use of the bus principle in the cabling freed from the constraint of having to specify during the system installation, which of the communication node takes over which function. Rather, with regard to the installation of each connected device is equivalent, and it will be decided only when operating the system, which function the respective device plays in the data transfer. The use of a time-division multiplexing method reduces the hardware complexity compared to frequency division multiplexing, in which a complex modulation and frequency conversion is necessary. Furthermore, a time division multiplex method with fixed time slots for each transmission channel is fully real-time capable, which is the case for other network systems, for example Using collision detection is not guaranteed.

In a further preferred embodiment, a basic conference block is implemented by a processor in conjunction with suitable operating software. The processor can continue to be used to provide a user interface, to encode data or to implement a transmission protocol. Such an arrangement has the advantage that an already existing processor can be used for the realization of the voice conference. Accordingly, no further than the already existing hardware for the realization of the voice conference is needed. In addition, the realization of a basic conference block using a processor over a hardwired solution offers the advantage that changes in functionality can be achieved solely by changing the software. This reduces development costs and enables a very universal and flexible system that can even be reconfigured at runtime.

In another preferred embodiment, a base conference block having a predetermined number of signal ports is formed by coupling a plurality of sub-conference blocks, each having fewer than the predetermined number of signal ports. This concept offers the advantage that a basic conference block can be modularly composed of several similar sub-conference blocks. Accordingly, it is sufficient to provide several small conference blocks. The result is a standardization of the required hardware or software, which brings a significant cost reduction. Furthermore, this concept is the basis for distributed system implementations.

A voice conference system comprises at least two basic conference blocks as described above and a device, via which the basic conference blocks are coupled together. In a preferred embodiment of a voice conference system, at least two of the basic conference blocks are spatially separated. This allows the realization of a distributed voice conference system.

In one embodiment, at least two of the base conference blocks are coupled via a wireless connection. This in turn allows a flexible extension of the voice conference system without installation effort.

In another embodiment, at least two of the basic conference blocks are coupled via a wired bus. In particular, it is advantageous if the wired bus is designed for the transmission of multiple voice channels in time division multiplex. Advantages of such an embodiment have already been explained in detail above.

In a further embodiment, the voice conference system has at least two terminals, wherein there is a duplex voice connection between all existing terminals. This has the advantage that all persons participating in the speech conference can hear and speak at the same time. Thus, the voice conference system simulates the presence of all participants in a common place in a good way.

In another embodiment of a voice conference system, one or more of the base conference blocks are part of a DECT base station. Furthermore, one or more of the terminals may be DECT mobile stations. Such an embodiment offers the advantage that a voice conference system can be integrated into an existing DECT system without much effort. Compatibility, in particular to existing devices is guaranteed. A digital and low-interference wireless transmission of the speech signal is also ensured. Furthermore, the manufacturer's effort to develop The hardware for the teleconferencing system is low because existing components can be reused with only minor changes.

In a further preferred embodiment, the DECT base stations and the DECT mobile stations are identical in terms of hardware. This offers the advantage of reducing the number of hardware variants in the system, which reduces manufacturing costs and enables flexible system reconfiguration at all times. The system features are defined only by software that can be easily replaced.

The present invention further provides two methods of operating a voice conference device or a voice conference system. In particular, the methods are implemented in the operation of a basic conference block, the realization of which can be implemented both in terms of hardware and by a microprocessor in conjunction with a suitable program. The advantage of the present method is that the processing of digitized speech signals necessary for realizing a speech conference can be carried out in a very simple and universal manner. Two methods are possible, wherein the first method according to claim 17 is advantageous if only a local voice conference is to be realized, or if the requirements for the temporal coordination of the voice signals are low. In a second method according to claim 18, the processing of the data supplied by local terminals and the processing of the sum signals supplied by other basic conference blocks is performed in separate steps. Such a method is advantageous if the requirements for the temporal coordination of the speech signals in a spatially widely distributed voice conference system are very high. In particular, in such a method, delays occurring on the transmission path can be almost completely compensated so that all the subscribers are independent of their current one Whereabouts and their distance from one another at the same time.

Fig. 1

einen Systemplan eines Ausführungsbeispiels eines Sprachkonferenzsystems;

Fig. 2

ein Blockschaltbild eines Ausführungsbeispiels eines Basis-Konferenzblocks mit maximal vier Teilnehmern;

Fig. 3

ein Blockschaltbild eines Ausführungsbeispiels eines Konferenzblocks mit sechs Teilnehmern;

Fig. 4

ein Blockschaltbild eines Ausführungsbeispiels eines Konferenzblocks mit neun Teilnehmern;

Fig. 5

ein Ablaufdiagramm eines Ausführungsbeispiels eines erfindungsgemäßem Verfahrens zum Betrieb eines Sprachkonferenzsystems.

Preferred embodiments of the present invention will be explained in more detail below with reference to the accompanying drawings. Show it:

Fig. 1

a system diagram of an embodiment of a voice conference system;

Fig. 2

a block diagram of an embodiment of a basic conference block with a maximum of four participants;

Fig. 3

a block diagram of an embodiment of a conference block with six participants;

Fig. 4

a block diagram of an embodiment of a conference block with nine participants;

Fig. 5

a flowchart of an embodiment of an inventive method for operating a voice conference system.

Fig. 1 shows a system diagram of an embodiment of a voice conference system, which is designated in its entirety by 10. The voice conference system comprises a master base station 12 and four similar slave base stations 14, two of which are shown here by way of example. Core of the master base station 12 is a conference block 20 with nine inputs. The core of a slave base station is a conference block 22 with six inputs. To the master base station 12 and to each of the slave base stations 14 are each a listener / station 24 and four handsets 26 are coupled, which can transmit digital audio signals via a bidirectional radio link. The listeners / stations 24 are connected here in an analogous way with the respective conference blocks, while the digitized voice signals are exchanged with the DECT handsets 26 ADPCM coded. ADPCM stands for adaptive differential pulse code modulation. The coupling of the listeners / stations 24 via devices 27 for converting analog signals into digital signals and vice versa, the coupling of the DECT handsets 26 via radio coupling means 29. The slave base stations 14 are connected to the master base station 12 via a serial Bus 30 connected. The provision and processing of the bus signals takes in each base stations 12,14 depending Fernkoppleinrichtung 31. The serial bus 30 comprises three lines, a data line 32 on which the data of four voice channels are transmitted in time division multiplex, a line 34 for the Bittakt and a line 36 for the frame clock.

Core of the master base station 12 is a conference block with nine subscribers 20, core of a slave base station 14 is a conference block 22 with six subscribers. Structure and operation of the conference blocks with nine participants and six participants 20, 22 are described below with reference to FIGS. 2-4, before finally explaining the operation of the entire voice conference system 10 of FIG. 1.

FIG. 2 shows a block diagram of an exemplary embodiment of a basic conference block with a maximum of four subscribers, which is designated by 50 in its entirety. The base conference block 50 for a maximum of four participants has four signal ports 52, each comprising an input 54 for a digitized speech signal and an output 56 for a digitized speech signal. The digital input signals to the four signal ports 52 are here designated A, B, C and D. Circuitry inside the base conference block 50 is designed to read in the input signals 54 from the four signal ports 52 and to output at the four outputs 56 the sum of the input signals 54 at the other signal ports 52, respectively. The Signal processing inside a basic conference block is completely digital. The base conference block thus realizes the core properties of a conference call. The input signal from the input 54 of a signal gate 52 is transmitted to the outputs 56 of the other signal ports 52, but not to the output 56 of the own signal gate 52.

FIG. 3 shows a block diagram of an exemplary embodiment of a six-party conference block, denoted 70 in its entirety. This is created by interconnecting two basic conference blocks 50a, 50b, each with four inputs. There are thus a total of six signal ports 72 available. Each base conference block 50a, 50b provides three external signal ports 72, 74, 76. Via a respective signal gate 52 of the two basic conference blocks 50a, 50b, the connection between these two is established. Again, it is ensured by the interconnection and the internal structure of the basic conference blocks, that at the output of each external signal port 72,74,76 the digitally formed sum of the input signals of the respective other signal ports 72,7.4,76 is applied.

FIG. 4 shows a block diagram of an exemplary embodiment of a conference block with nine subscribers, which is designated in its entirety by 90. This comprises four basic conference blocks 50c, 50d, 50e, 50f, each with four signal ports, and has nine external signal ports 92,94,96. A first base conference block 50c provides three external signal ports 92,94 and is connected at a signal port to another second base conference block 50d. The second base conference block 50d provides two external signal ports 92 and is connected at each one signal gate to the first base conference block 50c and a third base conference block 50e. This in turn provides two external signal ports 96 and is connected at its fourth signal port to a fourth base conference block 50f. The fourth basic conference block 50f represents in turn, two external signal ports 92 available. A signal gate of the fourth base conference block 50f is not connected. Thus, a total of nine external signal ports 92,94,96 are provided. Each output of an external signal gate has the sum of all inputs except its own signal gate.

The description of the base conference block 50 as well as the six-party extended conference blocks 70 and nine participants 90 shows that based on a small base conference block with a given number of signal ports, larger conference blocks with more signal ports can be assembled by appropriate circuitry. The essential properties of the signal blocks, in particular the fact that an input signal is transmitted to the outputs of all other signal ports, but not to the output of the own signal gate, are retained.

Based on the understanding of the base conference blocks 50 and the extended conference blocks 70, 90 with six participants and nine participants, the operation of the voice conference system 10 according to FIG. 1 will now be explained. In the present embodiment, both the master base station 12 and the slave base stations 14 are similarly constructed DECT base stations. The mobile parts 26 are DECT mobile stations in the present embodiment. The base stations can also be operated mobile. In the present embodiment, base stations and mobile stations are identical in terms of hardware and designed as a radio module. The difference is only in the software (firmware). For both base stations and mobile stations, any DECT derivative can be used in different frequency bands or with different frequency allocation techniques, e.g. B. also DECT in the 2.4 GHz band with frequency hopping method. The audio signals are transmitted between the base stations 12, 14 and the handsets 26 according to the DECT standard according to ADPCM coded over the air. A corresponding conversion into PCM-coded signals is possible and is carried out in the mobile units and in the base station.

Overall, the transmission via the air path is largely transparent, so that it can be assumed that the speech signal picked up by the respective mobile parts 26 is applied in digitized form at the inputs of the signal ports 92, 72, and that this is the case at the outputs of the signal ports 92, 72 present digitized voice signal is output via the speakers of the associated handsets 26. A bidirectional analog-to-digital converter circuit connected between the listeners 24 and the conference blocks 20, 22 ensures that the microphone signal of a respective local station at the DECT base station at the input of the associated signal gate 94, 74 in digital Form is present, and that the digitized speech signal is output at the output of the signal port 94,74 on the local speaker.

A distinction is made here between a local conference comprising only the mobile units 26 connected to the respective base station and the local listener / call station 24, as well as a global conference in which the local conferences of the respective base stations 12, 14 are combined to form an overall conference. Per base station several mobile stations can be operated in a local conference call. In a local conference, all participants directly connected to a base station can talk to each other. The connection between the base stations is not taken into account here at first. The local conference is achieved by adding the PCM coded voice signals from the handsets 26 and the local listener 24 in a conference block 20,22 so that at the output of a signal port 92,94,72,74 the sum of the Input signals of all other signal ports 92,94,72,74 of a conference block is present. The practical implementation of this calculation rule is relatively easy, in that the speech signals present in PCM-coded form are added digitally in the DECT processor of the base station 12, 14. The formation of a local conference thus does not require any additional circuitry. The maximum number of handsets 26 per base station 12,14 is hereby agreed by the system properties of the base station 12,14, z. From the maximum number of simultaneous radio links and the maximum number of simultaneously usable voice channels.

Based on the above-described procedure for forming a local conference, a global conference in which subscribers at all terminals 24, 26 connected to all base stations 12, 14 can speak to one another simultaneously can be set up. Here, the local conferences are advantageously combined into larger distributed conferences. The sub-conference signals are exchanged digitally between the base stations (PCM-coded). The base stations 12,14 are for this purpose via a wired serial bus 30, on which a plurality of logical channels are transmitted in time division multiplex. One of the identical base stations assumes the role of a master (master base station 12) by appropriate configuration. The master has the task of combining the sub-conferences of the other base stations (slave base stations 14). Furthermore, the master base station 12 takes over the synchronization of the serial bus 30. A logical bidirectional connection between the master base station 12 and each of the slave base stations 14 is established via the wired serial bus 30. The transmission in both directions and between the master 12 and the respective slave base station 14 by means of logical channels in the time division multiplex. The digitized speech signals are transmitted via this connection PCM-coded. Thus, the bus connection between the master base station 12 and the slave base stations 14 is transparent, ie equivalent to a direct bidirectional point-to-point connection between the master base station 12 and each of the slave base stations 14. All bus management tasks are performed by a remote link device 31.

The master base station thus essentially corresponds to a conference block 20 with nine subscribers, which is connected to four mobile audio / video stations 26, a local audio / video unit 24 and four other conference blocks 22. A slave base station essentially corresponds to a six-party conference block 22 which is connected to four mobile listeners / stations 26, a local listener / call station 24, and the base conference block 20.

Thus, the operation of the entire voice conference system is to be understood in detail as follows: In each handset 26, a wirelessly received input signal is output analogously to the loudspeaker of the respective listener / station. The microphone signal of the listener / station of each handset 26 is digitally transmitted to the corresponding base station 12,14. In the slave base stations, an input signal received by a local handset 26 over the air is output analogously to the local listener / station 24 and forwarded digitally to the other local mobiles 26 and to the master base station 12. The microphone signal at the local listener / station 24 of a slave base station 14 is digitally transmitted to the master base station 12 and to the local mobiles 26. In the slave base station 14, the signal coming from the master base station 12 is also output analogously at the listening / calling station 24 and digitally transmitted to the local mobile units 26.

In the master base station 12, an input signal received from a local handset 26 over the air is output analogously to the local listener 24 and to the other local mobiles 26 and to all slave base stations 14 digitally forwarded. The microphone signal at the local listener / station 24 of the master base station is digitally transmitted to all slave base stations 14 and to the local mobiles 26. In the master base station 12, the signal coming from a slave base station 14 is also output analogously at the listening / listening station 24 and digitally transmitted to the other slave base stations 14 and the local mobile units 26.

Thus, in the master base station 12, the local master base station conference is combined with the slave base station 14 local conferences. Thus, a substantial portion of the computational effort is outsourced to the slave base stations 14, in each of which already a local sub-conference is calculated. In particular, not all terminals 24, 26 must have a (direct or logical) connection to a central base station. The transmission of suitable sum signals as described above is sufficient. This significantly reduces the demands on the connection between the terminals and the central station.

It is obvious that the embodiment shown is not the only implementation of the present invention. Rather, the voice conference system described here can be changed in a wide context. For example, the number of handsets and the local listeners / stations can be changed. In the described embodiment, a master base station and four slave base stations are used. Each of the base stations is connected and synchronized with four handsets over a radio link. The mobile stations may therefore establish and release a radio link with audio signal transmission to the associated base station. Furthermore, the base station has a local listening / talking unit. The above-described voice conference system is thus suitable for setting up a voice conference comprising up to 25 participants.

Depending on requirements and technical possibilities, the number of mobile stations per base station can be changed. Furthermore, the local station at the base station can be omitted or more than one local station can be used. The number of slave base stations can be increased if, at the same time, the number of logical channels on the serial bus connecting the slave base stations to the master base station is increased. It can easily be seen that scalability at the local and global level constitutes a major advantage of the voice conference system according to the invention.

In another embodiment of the present invention, the connection between the master base station and the slave base stations may be realized in other ways. The wired serial bus can be replaced by any other connection, as long as it is ensured that voice transmission with sufficient quality and little time delay is possible. In particular, any high-speed bus system can be used. Furthermore, it is also conceivable to connect several base stations via a (preferably digital) public telephone network. The advantage of such a solution is that the voice conference system can be greatly expanded spatially. Furthermore, the connection between the master base station and the slave base stations can also be wireless. This has the advantage that the installation costs are significantly reduced. In principle, it is also conceivable to implement a voice conference system according to the present invention as an ad hoc system. That is, the configuration of the system and in particular the configuration of a wireless device as a terminal, as a master base station or as a slave base station is carried out automatically. This is possible since there is no difference in hardware between the terminal, master base station and slave base station. Accordingly, the hardware is very universally applicable and it Any configuration of the voice conference system can be easily achieved.

In another embodiment, the DECT base stations and DECT mobile stations may be replaced by base stations and mobile stations that obey a different standard. Even with the signal transmission to the mobile parts, the ADPCM coding can be replaced by any signal coding. The voice conference system according to the present invention is thus not bound to a particular communication standard.

Furthermore, it is possible to use the connection between several base stations not only for the transmission of digitized voice signals but also for the transmission of synchronization information. In particular, the transmission frequencies and the transmission cycles of the handsets in contact with the various base stations can be synchronized. This allows a more efficient use of the frequency resources.

For example, a first time division multiplex transmission between a first base conference block and the handsets in communication with the first base conference block and a second time division transmission between a second base conference block and the handsets in communication with the second base conference block coordinated temporally and / or frequency. This can be achieved that in the first time-division multiplex transmission only time slots are used, which are not used in the second time division multiplex transmission and vice versa. For this purpose, information about a frame structure of the time-division multiplex transmission between the base stations and the mobile parts can be transmitted between the base stations via the connection between the base stations, so that the base stations involved, for example, the same frame structure for communication with the mobile parts to be able to use. Thus, transmission cycles of the base stations and / or the mobile units in contact with different base stations can be coordinated.

The frame structure of the time-division multiplex transmission here describes, for example, time slots that are assigned to the individual base stations and / or mobile units or details of the assignment of time slots to base stations and / or mobile units. For example, the frame structure may provide that for a total of N mobiles the nth (1 <= n <= N) time slot of a frame is assigned to the communication between the nth handset and the associated base station (again a first portion of the n-th). the time slot of a transmission from the base station to the mobile part and a second part of the n-th time slot may be assigned to a transmission from the mobile part to the associated base station). A frame may also include further synchronization information.

The frame structure can thus be made known to all subscribers (base stations and preferably also mobile stations) through the connection between the base stations. Moreover, the frame structure of the time-division multiplex transmission can also be adapted dynamically to the number of existing mobile stations. A (data) transmission rate required by the individual mobile stations can also be taken into account when determining the frame structure.

For example, the described coordination of the frame structure facilitates the management of the movement of a particular mobile station. Namely, when the particular mobile station moves from the service area of a first base station to a service area of a second base station, using the base stations according to the invention with a remote switching facility for transmitting synchronization information, one time slot of the time-division multiplexed frame structure in which the first one Base station has communicated with the particular mobile station to transmit to the second base station for communication with the particular mobile station. The transmission of the corresponding synchronization information can be done via the remote coupling device. This is a "roaming", ie a transfer of a handset from one base station to another base station possible without replacing synchronization information with the handset. The synchronization is rather between the base stations using the remote coupling device.

However, it is also possible to use a fixed frame structure, which is distributed from a central base station to all other base stations via the remote coupling device, and which is optionally also transmitted to the mobile parts.

Moreover, it should be noted that not only information about the frame structure between the base stations can be exchanged via the remote switching device, but also that a temporal synchronization of the base stations can take place, so that all base stations simultaneously recognize a beginning of a frame.

The implementation of a basic conference block can also be greatly changed. The implementation is possible both by a processor in conjunction with a suitable program and in terms of hardware, for example in the form of a programmable logic module or an application-specific integrated circuit. Furthermore, the basic conference block can either be implemented as a whole or, as shown, divided into sub-blocks. Which form is cheaper depends essentially on the present form of realization and on the number of required signal ports.

In another embodiment, the hierarchical structure in which between master base station and Slave base station is to be canceled. For the functionality of a voice conference system according to the present invention, it is sufficient that a digitized voice signal can be transmitted from any conference block to any other conference block. If the transmission latency is low, the number of intermediate conference blocks will not matter. In particular, it is sufficient if there is a point-to-point connection between every two adjacent conference blocks. Such a system configuration corresponds to that shown in Fig. 4, wherein the basic conference blocks may well be spatially separated.

Likewise, in a further embodiment, a plurality of local sub-conferences may be formed. The calculation can be done both decentralized and in a master base station. In particular, it can also be determined at runtime of the system which station should act as the master base station.

5 shows a flow diagram of an embodiment of a method according to the invention for operating a voice conference system with two basic conference blocks. The method is executed in parallel in two basic conference blocks, wherein the method steps in the first conference block are denoted by 110 and the method steps in the second conference block by 112. In a first step, in both basic conference blocks digitized voice signals are read from the inputs of two or more signal ports coupled to terminals. In a second step, the signal values read in by the terminals are added in both conference blocks. The sum value is then output at a signal gate, which is connected to the other conference block. Accordingly, the sum value output by the respective other conference block is read in accordingly. Finally, in each conference block output values for the signal terminals connected to terminals are formed by summing the signal values read in by the respective other signal terminals connected to terminal devices and the summation values transmitted by the other basic conference block. These are output in a last step to the terminals.

The method according to the invention thus establishes a clear chronological order in the summation and the exchange of summation values between several conference blocks. The locally formed summation values must first be exchanged between the participating basic conference blocks before the final output value for the terminals can be calculated. Such a method is particularly necessary if the transmission time between individual conference blocks is long due to long transit times on the transmission link. The above-described method thus enables the signals of all the subscribers to be output simultaneously to a terminal, regardless of whether they are connected to the same conference block as the terminal or to a conference block remote from the room.

Thus, the present invention provides a voice conference system that enables a voice conference with very high voice quality. The audio signals are immediately converted at the microphones into digital signals, which are much more resistant to interference couplings than analog signals. Due to the decentralized formation of partial conferences, no complex central unit is needed. The base stations and the handsets may be identical in construction and differ only in the respective operating software. In particular, a voice conference system using the basic conference blocks according to the invention can be configured very flexibly and adapted to a large number of digital communication standards.

Claims (29)

1. A voice conference apparatus comprising:

   at least one base conference block (20, 22, 50) comprising three or more signal ports (52), each including an input (54) for a digitalized voice signal and an output (56) for a digitalized voice signal,

   wherein a base conference block (20, 22, 50) is formed such that, at a signal port (52), optionally a terminal (24, 26) or another base conference block (20, 22, 50) may be coupled using coupling means (27, 29, 31) for realizing a voice conference apparatus (10) extended with respect to the number of participants or in terms of space, and

   wherein a base conference block (20, 22, 50) is formed to enable a voice conference among the terminals (26) or base conference blocks (20, 22, 50) coupled thereto; and

   remote coupling means (31) formed to couple a base conference block (20, 22, 50) with at least one other spatially remote base conference block (20, 22, 50) in order to enable a spatially extended voice conference,

   wherein the remote coupling means (31) is further formed for a transmission of synchronization information in order to synchronize transmitting cycles of mobile units (PP1, PP2, PP3, PP4, PP5, PP6, PP7, PP8, PP17, PP18, PP19, PP20) in contact with various voice conference apparatuses (12, 14).
2. The voice conference apparatus according to claim 1, wherein the remote coupling means (31) is formed to couple a base conference block (20, 22, 50) with at least two other spatially remote base conference blocks (20, 22, 50) to which terminals (26) are coupled in order to enable a voice conference among the terminals (26) coupled to a total of at least three spatially distributed base conference blocks (20, 22, 50).
3. The voice conference apparatus according to claim 1 or 2, wherein a base conference block (20, 22, 50) is formed such that, at each output (56) of a signal port (52), a digitally created sum signal of the input signals at the inputs (54) of the respective other signal ports (52) is present in digital form.
4. The voice conference apparatus according to one of claims 1 to 3, wherein one of the coupling means includes a radio coupling means (29) formed such that a terminal (26) can be coupled to a base conference block (20, 22, 50) via a wireless connection.
5. The voice conference apparatus according to claim 4, wherein the radio coupling means (29) is formed such that it enables a wireless connection between the voice conference apparatus and a terminal (26) according to the DECT standard or a standard based thereon.
6. The voice conference apparatus according to one of claims 1 to 5, wherein a talk-listen unit (24) is coupled to the base conference block (20, 22, 50) via means (27) for the conversion of analog signals to digital signals and means for the conversion of digital signals to analog signals.
7. The voice conference apparatus according to one of claims 1 to 6, wherein the remote coupling means (31) is formed such that at least one other base conference block (22, 50) may be coupled to a first base conference block (20, 50) via a wireless connection.
8. The voice conference apparatus according to one of claims 1 to 7, wherein the remote coupling means (31) is formed such that at least one other base conference block (22, 50) may be coupled to a first base conference block (20, 50) via a wire bus (30).
9. The voice conference apparatus according to claim 8, wherein the wire bus (30) is formed for a transmission of several voice channels in time-division multiplex.
10. The voice conference apparatus according to one of claims 1 to 9, wherein the base conference block (20, 22, 50) is realized by a processor in connection with suitable operational software.
11. The voice conference apparatus according to claim 10, wherein the processor is simultaneously used for at least one of the following tasks:

    provision of a user interface,

    coding of data,

    realization of a transmission protocol.
12. The voice conference apparatus according one of claims 1 to 11, wherein a base conference block (20, 22, 70, 90) with a predetermined number of signal ports (72, 92) is formed by coupling several sub-conference blocks (50), each comprising less than the predetermined number of signal ports (52),
    wherein a sub-conference block (50) is formed such that, at a signal port (52), optionally a terminal (24, 26), another sub-conference block (50), or a base conference block (20, 22, 70, 90) may be coupled using coupling means (27, 29, 31).
13. A voice conference system comprising:

    at least two spatially separate base conference blocks (20, 22) comprising three or more signal ports (72, 92), each including an input for a digitalized voice signal and an output for a digitalized voice signal,

    wherein a base conference block (20, 22) is formed such that, at a signal port (72, 92), optionally a terminal (24, 26) or another base conference block (20, 22) may be coupled using coupling means (27, 29, 31) for the realization of a voice conference apparatus extended with respect to the number of participants or in terms of space; and

    remote coupling means (30, 31) via which the base conference blocks are coupled to one another,

    wherein the remote coupling means (31) is further formed for a transmission of synchronization information in order to synchronize transmitting cycles of mobile units (PP1, PP2, PP3, PP4, PP5, PP6, PP7, PP8, PP17, PP18, PP19, PP20) in contact with various base conference blocks (12, 14).
14. The voice conference system according to claim 13, including at least three base conference blocks (20, 22) that are separated in terms of space.
15. The voice conference system according to claim 13 or 14, wherein at least two of the base conference blocks (20, 22) are coupled via a wireless connection.
16. The voice conference system according to one of claims 13 to 15, wherein at least two of the base conference blocks (20, 22) are coupled via a wire bus (30).
17. The voice conference system according to claim 16, wherein the wire bus (30) is formed for the transmission of several voice channels in time-division multiplex.
18. The voice conference system according to one of claims 13 to 17, wherein the voice conference system has at least two terminals (24, 26), wherein a duplex voice link exists among all terminals.
19. The voice conference system according to one of claims 13 to 18, wherein one or more of the terminals (26) are coupled to a base conference block via a wireless connection.
20. The voice conference system according to claim 19, wherein the wireless connection to one or more terminals (26) is effected according to the DECT standard or a standard based thereon.
21. The voice conference system according to one of claims 13 to 20, wherein one or more of the base conference blocks (20, 22) are part of a DECT base station (12, 14).
22. The voice conference system according to one of claims 13 to 21, wherein one or more of the terminals are DECT mobile units (26).
23. The voice conference system according to claims 21 and 22, wherein the DECT base stations (12, 14) and the DECT mobile units (26) match in hardware.
24. The voice conference system according to one of claims 13 to 23, wherein a talk-listen unit (24) is coupled to a base conference block (20, 22, 50) via means (27) for the conversion of analog signals to digital signals and means (27) for the conversion of digital signals to analog signals.
25. The voice conference system according to one of claims 13 to 24, wherein a base conference block (20, 22, 50) is formed such that, at each output (56) of a signal port (52), a digitally created sum signal of the input signals of the respective other signal ports (52) is present in digital form.
26. The voice conference system according to one of claims 13 to 25, wherein a base conference block (20, 22, 50) is realized by a processor in connection with suitable operational software.
27. The voice conference system according to claim 26, wherein the processor is used simultaneously for at least one of the following tasks:

    provision of a user interface,

    coding of data,

    realization of a transmission protocol.
28. The voice conference system according to one of claims 13 to 27, wherein a base conference block (20, 22, 70, 90) with a predetermined number of signal ports (72, 92) is formed by coupling several subconference blocks (50), each comprising less than the predetermined number of signal ports (52),
    wherein a subconference block (50) is formed such that, at each signal port (52), optionally a terminal (24, 26), another subconference block (50) or a base conference block (20, 22, 70, 90) may be coupled using coupling means.
29. A method for operating a distributed voice conference system with two base conference blocks (20, 22) coupled to each other via a signal port each according to one of claims 13 to 28, comprising:

    reading the digitalized voice signals from the inputs of the two or more signal ports of a first base conference block (20) which are coupled to the terminals (24, 26);

    reading the digitalized voice signals from the inputs of the two or more signal ports of a second base conference block (22) which are coupled to the terminals (24, 26);

    calculating a first sum value from the read digitalized voice signals of the first base conference block (20);

    calculating a second sum value from the read digitalized voice signals of the second base conference block (22);

    outputting the first sum value at the output of the signal port of the first base conference block (20), which is coupled to a signal port of the second base conference block (22);

    outputting the second sum value at the output of the signal port of the second base conference block (22), which is coupled to a signal port of the first base conference block (20);

    reading the second sum value from the input of the signal port of the first base conference block (20), which is coupled to a signal port of the second base conference block (22);

    reading the first sum value from the input of the signal port of the second base conference block (22), which is coupled to a signal port of the first base conference block (20);

    calculating output values at the outputs of the signal ports of the first base conference block (20) connected to a terminal (24, 26) by the summation of the values read from the inputs of the respective other signal ports of the first base conference block (20) including the second sum value transmitted from the second base conference block (22);

    calculating output values at the outputs of the signal ports of the second base conference block (22) connected to a terminal (24, 26) by the summation of the values read from the inputs of the respective other signal ports of the second base conference block (22) including the first sum value transmitted from the first base conference block (20); and

    outputting the output values at the outputs of the signal ports of the two base conference blocks (20, 22), which are coupled to terminals (24, 26).

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